multi dispositivo con envío independientes por capa. está sucio con

trozos sn usar y statics, pero funciona.

docs/LibreMediaServer_Audio.hed

@@ -1,93 +1,99 @@
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-<EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>閦悌洮隆豆此
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+<EFBFBD>丕厉徉牠衔吞耸耸燮拍寐晾俾
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+徽<EFBFBD>地<EFBFBD>敝<EFBFBD><EFBFBD>Ш垢范刀碑毕瓮趟怂<EFBFBD>
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+<EFBFBD><EFBFBD><EFBFBD>螯撱摐煘潨洑洑嫋晹搾憮<EFBFBD>
+吭逸迅凶苴本<EFBFBD>烤郊垢ジ范荡吵嘤
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+濏<EFBFBD><EFBFBD>殝殯唴剝倎仢倽湜殭槜膲
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+忉圜怙湩<EFBFBD>鑻┆瘁铖饗軕寢寠攪唴厽倎€煘潨泦
+櫂棡墕笪蕾控蝠<EFBFBD><EFBFBD>牒沪アЬ<EFBFBD>疙〖缓袱范荡巢毕<EFBFBD>
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+<EFBFBD><EFBFBD>骐槴Ъ鋮<EFBFBD>ぺ虧钰詻攪唽偀倎咽倽湜泤槜枙敁拺<EFBFBD>
+截麴<EFBFBD>淮Еゥ竣　竣疥缓垢贩┐巢坝瓮倘<EFBFBD>
+劬灇惼艳晾哌淋圹倌种赵右诱箢盱牿殍玟<EFBFBD>
+鰰唤叠灬<EFBFBD><EFBFBD>澉鲷麸篌饛帊實枆垏嚈剝個<EFBFBD>
+嶈睦螝媱棖晻彃憫<EFBFBD><EFBFBD><EFBFBD><EFBFBD>┄ェ工＂牸烤郊<EFBFBD>
+ㄏ珉忄霭牠衔屯资缮勤睦寐晾挢淋圳倌字兆<EFBFBD>
+困彸焊<EFBFBD><EFBFBD>桤驿<EFBFBD>忄嚆<EFBFBD><EFBFBD><EFBFBD><EFBFBD>牾趔蝽饛帍<EFBFBD>
+欬衷佑菂憺亐煙仠洓檮柪晹搾搾钞<EFBFBD><EFBFBD>┄Ё<EFBFBD>
+睹箐汴<EFBFBD>蓟焊し洞疮氨衔吞藶杖瞧拓寐聯<EFBFBD>
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+凁<EFBFBD><EFBFBD>浼贼<EFBFBD>沪イ⒕<EFBFBD>揪【缓垢范悒巢鼻彝趟险
+惹婆拿铝<EFBFBD>
+咿蒈圳儇酥赵右研矧盱腙殍珂<EFBFBD>汊徉<EFBFBD><EFBFBD>帑<EFBFBD><EFBFBD>鲺麸蝰饛帊寢枆垏唴剝倽€煘潨洑檮棖晹搾憪钞<EFBFBD><EFBFBD>┄Шイ＂<EFBFBD>烤〖缓垢范胆巢毕瓮趟稚惹婆拿螺肋掭苒谫淖终杂已畜铐祀觊桤<EFBFBD>溷忉<EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD><EFBFBD>鲺梵蝰饛帊寳妷垏唴剝瀬€煘潨洑厴棖晹搾憣<EFBFBD><EFBFBD><EFBFBD>┄沪イ＂<EFBFBD>竣
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libremediaserver-audio.pro

@@ -8,9 +8,11 @@ HEADERS += src/libremediaserver-audio.h \
     src/dmxwidget.h \
     src/filterbankwidget.h \
     src/libremediaserver-audio-gui.h \
+    src/ma_writer_node.h \
     src/main.h \
-    src/miniaudio.h \
     src/medialibrary.h \
+    src/miniaudio.h \
+    src/ma_writer_node.h \
     src/miniaudioengine.h \
     src/olathread.h \
     src/audiolayerwidget.h \
@@ -26,10 +28,11 @@ SOURCES += src/main.cpp \
     src/dmxwidget.cpp \
     src/filterbankwidget.cpp \
     src/libremediaserver-audio-gui.cpp \
-    src/miniaudio.c \
     src/libremediaserver-audio.cpp \
     src/medialibrary.cpp \
+    src/miniaudio.c \
     src/miniaudioengine.cpp \
+    src/ma_writer_node.c \
     src/olathread.cpp \
     src/audiolayerwidget.cpp \
     src/audiowidget.cpp \

src/defines.h

@@ -1,15 +1,18 @@
 #ifndef DEFINES_H
 #define DEFINES_H
 
-#define VERSION "LibreMediaServerAudio v0.2.0 Antigona"
-#define COPYRIGHT "(C) 2014-2024 Santi Noreña <lms@criptomart.net>"
-#define LICENSE "GPL 3 Licensed. See LICENSE.txt."
-#define DEFAULT_FILE "lms-audio.xlm"
-#define MAX_LAYERS 4
-#define MAX_AUDIODEVICES 8
-#define UI_REFRESH_TIME 100
-#define FADE_TIME 25 // DMX Frame time, 40 fps, avoid clicks
-#define FILTER_CHANNELS 13 // number of dmx channels dedicated to filters by layer
+#define VERSION             "LibreMediaServerAudio v0.2.0 Antigona"
+#define COPYRIGHT           "(C) 2014-2024 Santi Noreña <lms@criptomart.net>"
+#define LICENSE             "GPL3 Licensed. See LICENSE.txt."
+#define DEFAULT_FILE        "lms-audio.xlm"
+#define MAX_LAYERS          4
+#define MAX_AUDIODEVICES    8
+#define FORMAT              ma_format_f32   /* Must always be f32. */
+#define CHANNELS            2
+#define SAMPLE_RATE         48000
+#define UI_REFRESH_TIME     100
+#define FADE_TIME           25 // DMX Frame time, 40 fps, avoid clicks
+#define FILTER_CHANNELS     13 // number of dmx channels dedicated to filters by layer
 
 struct dmxSetting {
     int address;
@@ -18,7 +21,7 @@ struct dmxSetting {
     int audioDevice;
 };
 
-enum class Status
+enum Status
 {
     Stopped,
     Paused,
@@ -30,6 +33,15 @@ enum class Status
     PlayingFolderRandom
 };
 
+enum Slider
+{
+    Volume,
+    Pan,
+    Pitch,
+    Bypass
+};
+
+#ifdef __cplusplus
 constexpr const char* statusToString(Status e) noexcept
 {
     switch (e)
@@ -45,27 +57,6 @@ constexpr const char* statusToString(Status e) noexcept
         default: return "--++--";
     }
 }
-/*
-static const char* StatusStr[] =
-{
-    "Stop",
-    "Pause",
-    "Play One",
-    "Play One Loop",
-    "Iddle",
-    "Play Folder",
-    "Play Folder Loop",
-    "Play Folder Rand",
-    0x0
-};*/
-
-enum Slider
-{
-    Volume,
-    Pan,
-    Pitch,
-    Bypass
-};
 
 #include <QString>
 struct layerData {
@@ -81,4 +72,5 @@ struct layerData {
     unsigned int universe;
     int device;
 };
+#endif // __cplusplus
 #endif // DEFINES_H

src/dmxPersonality.h

@@ -23,9 +23,10 @@
 #define HIGH_FREQ 19
 #define HIGH_Q 20
 #define HIGH_GAIN 21
-#define SEND1 22
-#define SEND2 23
-#define LAYER_CHANNELS 24
+#define FILTER_BANK_GAIN 22 // not implemented yet
+#define SEND1 23
+#define SEND2 24
+#define LAYER_CHANNELS 25
 
 constexpr const char* dmxChannelToString(int e) noexcept
 {
@@ -43,6 +44,7 @@ constexpr const char* dmxChannelToString(int e) noexcept
         case HIGH_FREQ: return "High Cutoff Frec";
         case HIGH_Q: return "High Slope";
         case HIGH_GAIN: return "High Gain";
+        case FILTER_BANK_GAIN: return "Post Filters Gain";
     default: return "++--++--++";
     }
 }

src/filterbankwidget.cpp

@@ -10,10 +10,10 @@ FilterBankWidget::FilterBankWidget(QWidget *parent)
     layout->setAlignment(Qt::AlignHCenter);
     layout->setContentsMargins(0, 2, 0, 2);
     layout->setSpacing(0);
-    this->setStyleSheet("border: 1px solid #5a4855;"
+    this->setStyleSheet("border: 2px solid #5a4855;"
                 "margin: 0px;"
-                "margin-top: 2px;"
-                "margin-bottom: 2px;"
+                "margin-top: 3px;"
+                "margin-bottom: 3px;"
                 "background-color: #383034;"
                 );
     for (int i = 0; i < 13; i++) {
@@ -25,8 +25,13 @@ FilterBankWidget::FilterBankWidget(QWidget *parent)
     QVBoxLayout *master = new QVBoxLayout;
     fb[0]->setRange(0, 500);
     m_bypass = new QCheckBox;
-    connect(m_bypass, SIGNAL(stateChanged(int)), this, SLOT(bypassChanged(int)));
     master->addWidget(m_bypass);
+    m_bypass->setText("Bypass");
+    m_bypass->setStyleSheet("QCheckBox { border: 2px solid #2a0825;"
+                "margin: 0px;"
+                "background-color: #885074;"
+                "font-size: 7px;}");
+    connect(m_bypass, SIGNAL(stateChanged(int)), this, SLOT(bypassChanged(int)));
     master->addWidget(fb[0]);
     layout->addLayout(master);
     for (int i = 1; i < 13;) {

src/libremediaserver-audio-gui.cpp

@@ -35,7 +35,7 @@ libreMediaServerAudioUi::libreMediaServerAudioUi(QWidget *parent)
     topWidget->setContentsMargins(0, 0, 0, 0);
     addDockWidget(Qt::TopDockWidgetArea, topWidget);
     connect(ui.actionLaunch_OLA_Setup, SIGNAL(triggered()), this, SLOT(olasetup()));
-    this->setContentsMargins(5, 5, 5, 5);
+    this->setContentsMargins(3, 3, 3, 3);
     this->setStyleSheet(
                 "color: white;"
                 "background-color: #4f4048;"

src/libremediaserver-audio.cpp

@@ -140,7 +140,7 @@ void libreMediaServerAudio::dmxInput(int layer, int channel, int value)
             m_played.append(m_ola->getValue(layer, DMX_FILE));
         }
 #endif
-    } else if (channel >= HP_FREQ && channel <= HIGH_GAIN) {
+    } else if (channel >= HP_FREQ) {
         m_mae.filterParamChanged(layer, m_dmxSettings.at(layer).audioDevice, channel, value);
 #ifndef NOGUI
         if (m_ui) {

src/ma_writer_node.c

@@ -0,0 +1,178 @@
+#include "ma_writer_node.h"
+#include "miniaudio.c"
+MA_API ma_writer_node_config ma_writer_node_config_init(ma_uint32 channels, ma_uint32 bufferSizeInFrames, ma_pcm_rb *rb)
+{
+    ma_writer_node_config config;
+
+    MA_ZERO_OBJECT(&config);
+    config.nodeConfig = ma_node_config_init();
+    config.channels   = channels;
+    config.bufferSizeInFrames = bufferSizeInFrames;
+    config.pBuffer = rb;
+
+    return config;
+}
+
+static void ma_writer_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
+{
+    ma_writer_node* pWriteNode = (ma_writer_node*)pNode;
+
+    MA_ASSERT(pWriteNode != NULL);
+    MA_ASSERT(ma_node_get_input_bus_count(&pWriteNode->baseNode) == 2);
+
+    if (*pFrameCountIn > 0) {
+        void *pWriteBuffer = NULL;
+        ma_pcm_rb_acquire_write(pWriteNode->pBuffer, pFrameCountIn, &pWriteBuffer);
+        if (pWriteBuffer != NULL) {
+            ma_copy_pcm_frames(pWriteBuffer, ppFramesIn[1], *pFrameCountIn, ma_format_f32, pWriteNode->channels);
+            ma_pcm_rb_commit_write(pWriteNode->pBuffer, *pFrameCountIn);
+        }
+    }
+    //*pFrameCountOut = 0;
+    ma_copy_pcm_frames(ppFramesOut[0], ppFramesIn[0], *pFrameCountOut, ma_format_f32, pWriteNode->channels);
+}
+
+static ma_node_vtable g_ma_writer_node_vtable =
+{
+    ma_writer_node_process_pcm_frames,
+    NULL,
+    2,
+    1,
+    0
+//    MA_NODE_FLAG_CONTINUOUS_PROCESSING
+//    MA_NODE_FLAG_SILENT_OUTPUT
+};
+
+MA_API ma_result ma_writer_node_init(ma_node_graph* pNodeGraph, const ma_writer_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_writer_node* pWriteNode)
+{
+    ma_result result;
+    ma_node_config baseConfig;
+    ma_uint32 inputChannels[2];     // Equal in size to the number of input channels specified in the vtable.
+    ma_uint32 outputChannels[1];
+
+    if (pWriteNode == NULL || pConfig == NULL || pConfig->pBuffer == NULL \
+                           || (pConfig->channels > MA_MAX_NODE_BUS_COUNT) ) {
+        return MA_INVALID_ARGS;
+    }
+
+    MA_ZERO_OBJECT(pWriteNode);
+    inputChannels[0]  = pConfig->channels;
+    inputChannels[1]  = pConfig->channels;
+    outputChannels[0] = pConfig->channels;
+    baseConfig = pConfig->nodeConfig;
+    baseConfig.vtable          = &g_ma_writer_node_vtable;
+    baseConfig.pInputChannels  = inputChannels;
+    baseConfig.pOutputChannels = outputChannels;
+
+    result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pWriteNode->baseNode);
+    if (result != MA_SUCCESS) {
+        return result;
+    }
+	pWriteNode->bufferSizeInFrames = pConfig->bufferSizeInFrames;
+	pWriteNode->pBuffer = pConfig->pBuffer;
+	pWriteNode->channels = pConfig->channels;
+	return MA_SUCCESS;
+}
+
+MA_API void ma_writer_node_uninit(ma_writer_node* pWriteNode, const ma_allocation_callbacks* pAllocationCallbacks)
+{
+    ma_node_uninit(&pWriteNode->baseNode, pAllocationCallbacks);
+}
+
+/*
+ * Data Source Ring Buffer
+ */
+
+
+ma_result ma_data_source_rb_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
+{
+     ma_data_source_rb* ds = (ma_data_source_rb*)pDataSource;
+
+    ma_uint32 pcmFramesAvailableInRB = 0;
+    ma_uint32 pcmFramesProcessed = 0;
+    // lo mismo que en el callback, va el doble de rápido y con glitches.
+    while (pcmFramesProcessed < frameCount) {
+        pcmFramesAvailableInRB = ma_pcm_rb_available_read(ds->rb);
+        if (pcmFramesAvailableInRB == 0) {
+            break;
+        }
+        ma_uint32 framesToRead = frameCount - pcmFramesProcessed;
+        if (framesToRead > pcmFramesAvailableInRB) {
+            framesToRead = pcmFramesAvailableInRB;
+        }
+        void* pReadBuffer = NULL;
+        ma_pcm_rb_acquire_read(ds->rb, &framesToRead, &pReadBuffer);
+        if (pReadBuffer != NULL) {
+            ma_copy_pcm_frames(pFramesOut, pReadBuffer, framesToRead, ma_format_f32, 2);
+            ma_pcm_rb_commit_read(ds->rb, framesToRead);
+            pcmFramesProcessed += framesToRead;
+        }
+        else {
+            break;
+        }
+    }
+    *pFramesRead += pcmFramesProcessed;
+    return MA_SUCCESS;
+}
+
+ma_result ma_data_source_rb_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
+{
+    (void)pDataSource;
+    (void)frameIndex;
+    return MA_NOT_IMPLEMENTED;
+}
+
+ma_result ma_data_source_rb_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
+{
+    (void)pDataSource;
+    *pFormat = ma_format_f32;
+    *pChannels = 2;
+    *pSampleRate = ma_standard_sample_rate_48000;
+    return MA_SUCCESS;
+}
+
+ma_result ma_data_source_rb_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
+{
+    (void)pDataSource;
+    *pCursor  = 0;
+    return MA_NOT_IMPLEMENTED;
+}
+
+ma_result ma_data_source_rb_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
+{
+    (void)pDataSource;
+    *pLength  = 0;
+    return MA_NOT_IMPLEMENTED;
+}
+
+ma_data_source_vtable g_ma_data_source_rb_vtable =
+{
+    ma_data_source_rb_read,
+    ma_data_source_rb_seek,
+    ma_data_source_rb_get_data_format,
+    ma_data_source_rb_get_cursor,
+    ma_data_source_rb_get_length
+};
+
+ma_result ma_data_source_rb_init(ma_data_source_rb* pMyDataSource, ma_pcm_rb *ringBuffer)
+{
+    ma_result result;
+    ma_data_source_config baseConfig;
+
+    baseConfig = ma_data_source_config_init();
+    baseConfig.vtable = &g_ma_data_source_rb_vtable;
+
+    result = ma_data_source_init(&baseConfig, &pMyDataSource->base);
+    if (result != MA_SUCCESS) {
+        return result;
+    }
+
+    pMyDataSource->rb = ringBuffer;
+
+    return MA_SUCCESS;
+}
+
+void ma_data_source_rb_uninit(ma_data_source_rb* pMyDataSource)
+{
+    ma_data_source_uninit(&pMyDataSource->base);
+}

src/ma_writer_node.h

@@ -0,0 +1,52 @@
+/* Include ma_writer_node.h after miniaudio.h */
+#ifndef ma_writer_node_h
+#define ma_writer_node_h
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "miniaudio.h"
+
+typedef struct
+{
+    ma_node_config nodeConfig;
+    ma_uint32 channels;
+    ma_uint32 bufferSizeInFrames;
+    ma_pcm_rb *pBuffer;
+} ma_writer_node_config;
+
+MA_API ma_writer_node_config ma_writer_node_config_init(ma_uint32 channels, ma_uint32 bufferSizeInFrames, ma_pcm_rb *rb);
+
+typedef struct
+{
+    ma_node_base baseNode;
+    ma_uint32 bufferSizeInFrames;
+    ma_pcm_rb *pBuffer;
+	ma_uint32 channels;
+} ma_writer_node;
+
+MA_API ma_result ma_writer_node_init(ma_node_graph* pNodeGraph, const ma_writer_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_writer_node* pWriteNode);
+MA_API void ma_writer_node_uninit(ma_writer_node* pWriteNode, const ma_allocation_callbacks* pAllocationCallbacks);
+/**
+ * data source ring buffer
+ */
+
+typedef struct
+{
+    ma_data_source_base base;
+    ma_pcm_rb *rb;
+} ma_data_source_rb;
+
+ma_result ma_data_source_rb_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
+ma_result ma_data_source_rb_seek(ma_data_source* pDataSource, ma_uint64 frameIndex);
+ma_result ma_data_source_rb_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
+ma_result ma_data_source_rb_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor);
+ma_result ma_data_source_rb_get_length(ma_data_source* pDataSource, ma_uint64* pLength);
+ma_result ma_data_source_rb_init(ma_data_source_rb* pMyDataSource, ma_pcm_rb *ringBuffer);
+void ma_data_source_rb_uninit(ma_data_source_rb* pMyDataSource);
+
+#ifdef __cplusplus
+}
+#endif
+#endif  /* ma_writer_node_h */

src/miniaudioengine.cpp

@@ -4,16 +4,48 @@
 #define BIAS                0.99f
 #define FILTER_ORDER        3
 
+static ma_pcm_rb aux1Buffer;
+static ma_data_source_node g_dataSupplyNode;
+static ma_data_source_rb g_dataSourceRB;
+
 MiniAudioEngine::MiniAudioEngine() {}
 
 void MiniAudioEngine::audioDataCallback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    (void)pInput;
     ma_result result;
+
     result = ma_engine_read_pcm_frames((ma_engine*)pDevice->pUserData, pOutput, frameCount, NULL);
     if (result != MA_SUCCESS) {
         cout << "Error " << result << ": error audio callback.";
     }
+    (void)pInput;
+}
+
+void MiniAudioEngine::audioDataCallback1(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
+{
+    (void)pDevice;
+    ma_result result;
+
+    ma_uint32 pcmFramesAvailableInRB = 0;
+    ma_uint32 pcmFramesProcessed = 0;
+    while (pcmFramesProcessed < frameCount) {
+        pcmFramesAvailableInRB = ma_pcm_rb_available_read(&aux1Buffer);
+        if (pcmFramesAvailableInRB == 0) {
+            break;
+        }
+        ma_uint32 framesToRead = frameCount - pcmFramesProcessed;
+        if (framesToRead > pcmFramesAvailableInRB) {
+            framesToRead = pcmFramesAvailableInRB;
+        }
+        void* pReadBuffer = NULL;
+        ma_pcm_rb_acquire_read(&aux1Buffer, &framesToRead, &pReadBuffer);
+        if (pReadBuffer != NULL) {
+            ma_copy_pcm_frames(pOutput, pReadBuffer, framesToRead, FORMAT, CHANNELS);
+            ma_pcm_rb_commit_read(&aux1Buffer, framesToRead);
+            pcmFramesProcessed += framesToRead;
+        }/* else { break; }*/
+    }
+    (void)pInput;
 }
 
 void MiniAudioEngine::stopEngine()
@@ -23,6 +55,7 @@ void MiniAudioEngine::stopEngine()
     }
     for (uint i = 0; i < m_devicesSelected; i++) {
         for (uint j = 0; j < m_layersQty; j++) {
+            ma_node_uninit(&m_filterBank[i][j].input, NULL);
             ma_hpf_node_uninit(&m_filterBank[i][j].hpf, NULL);
             ma_loshelf_node_uninit(&m_filterBank[i][j].loshelf, NULL);
             ma_peak_node_uninit(&m_filterBank[i][j].mLow, NULL);
@@ -30,6 +63,8 @@ void MiniAudioEngine::stopEngine()
             ma_hishelf_node_uninit(&m_filterBank[i][j].hishelf, NULL);
             ma_splitter_node_uninit(&m_filterBank[i][j].output, NULL);
         }
+        //ma_writer_node_uninit(&m_sendToAux[0], NULL);
+        //ma_pcm_rb_uninit(&aux1Buffer);
         ma_engine_uninit(&m_engine[i]);
         ma_device_uninit(&m_device[i]);
     }
@@ -60,56 +95,55 @@ bool MiniAudioEngine::startEngine(uint layers)
 ma_result MiniAudioEngine::createFilterBank(int id, uint layer)
 {
     ma_result result;
-
     ma_node_graph *ng = ma_engine_get_node_graph(&m_engine[id]);
     ma_node *endpoint = ma_engine_get_endpoint(&m_engine[id]);
-    filterBank *fb = &m_filterBank[id][layer];
+    filterBank *fb    = &m_filterBank[id][layer];
 
     ma_splitter_node_config splitterConfig = ma_splitter_node_config_init(CHANNELS);
     result = ma_splitter_node_init(ng, &splitterConfig, NULL, &fb->input);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to initialize input node." << endl;
+        cout << "ERROR " << result << ": Failed to init input node." << endl;
         return result;
     }
 
     fb->hpfConfig = ma_hpf_node_config_init(CHANNELS, SAMPLE_RATE, 16, FILTER_ORDER);
     result = ma_hpf_node_init(ng, &fb->hpfConfig, NULL, &fb->hpf);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to initialize high pass filter node." << endl;
+        cout << "ERROR " << result << ": Failed to init high pass filter node." << endl;
         return result;
     }
 
     fb->loshelfConfig = ma_loshelf_node_config_init(CHANNELS, SAMPLE_RATE, 0.0f, 1.0f, 30);
     result = ma_loshelf_node_init(ng, &fb->loshelfConfig, NULL, &fb->loshelf);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to initialize low pass filter node." << endl;
+        cout << "ERROR " << result << ": Failed to init low pass filter node." << endl;
         return result;
     }
 
     fb->mLowConfig = ma_peak_node_config_init(CHANNELS, SAMPLE_RATE, 0.0, 4.0, 200); // double gainDB, double q, double frequency);
     result = ma_peak_node_init(ng, &fb->mLowConfig, NULL, &fb->mLow);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to initialize peak low filter node." << endl;
+        cout << "ERROR " << result << ": Failed to init peak low filter node." << endl;
         return result;
     }
 
     fb->mHighConfig = ma_peak_node_config_init(CHANNELS, SAMPLE_RATE, 0.0, 0.0, 600); // double gainDB, double q, double frequency);
     result = ma_peak_node_init(ng, &fb->mHighConfig, NULL, &fb->mHigh);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to initialize peak high filter node." << endl;
+        cout << "ERROR " << result << ": Failed to init peak high filter node." << endl;
         return result;
     }
 
     fb->hishelfConfig = ma_hishelf_node_config_init(CHANNELS, SAMPLE_RATE, 0.0f, 1.0f, 20000);
     result = ma_hishelf_node_init(ng, &fb->hishelfConfig, NULL, &fb->hishelf);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to initialize hi shelf filter node." << endl;
+        cout << "ERROR " << result << ": Failed to init hi shelf filter node." << endl;
         return result;
     }
 
     result = ma_splitter_node_init(ng, &splitterConfig, NULL, &fb->output);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to initialize output node." << endl;
+        cout << "ERROR " << result << ": Failed to init output node." << endl;
         return result;
     }
 
@@ -120,7 +154,7 @@ ma_result MiniAudioEngine::createFilterBank(int id, uint layer)
     }
     result = ma_node_attach_output_bus(&fb->input, 1, &fb->output, 0);
     if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to attach high pass pass filter node." << endl;
+        cout << "ERROR " << result << ": Failed to attach bypass connection." << endl;
         return result;
     }
     ma_node_set_output_bus_volume(&fb->input, 1, 0.0f);
@@ -149,10 +183,17 @@ ma_result MiniAudioEngine::createFilterBank(int id, uint layer)
         cout << "ERROR " << result << ": Failed to attach high shelf filter node." << endl;
         return result;
     }
-    result = ma_node_attach_output_bus(&fb->output, 0, endpoint, 0);
-    if (result != MA_SUCCESS) {
-        cout << "ERROR " << result << ": Failed to attach output node to engine." << endl;
-        return result;
+if (id == 0) {
+        result = ma_node_attach_output_bus(&fb->output, 0, &m_sendToAux[id], 0);
+        if (result != MA_SUCCESS) {
+            cout << "ERROR " << result << ": Failed to attach output node to engine." << endl;
+            return result;
+        }
+        result = ma_node_attach_output_bus(&fb->output, 1, &m_sendToAux[id], 1);
+        if (result != MA_SUCCESS) {
+            cout << "ERROR " << result << ": Failed to attach output node to aux send 1." << endl;
+            return result;
+        }
     }
     return result;
 }
@@ -161,6 +202,27 @@ ma_result MiniAudioEngine::setNodeGraph(int id) {
     ma_result result = MA_SUCCESS;
     uint i = 0;
 
+    if (id == 0) {
+        ma_node_graph *ng = ma_engine_get_node_graph(&m_engine[id]);
+        size_t sizeInFrames = SAMPLE_RATE; // ma_get_bytes_per_frame(FORMAT, CHANNELS);
+        result = ma_pcm_rb_init(FORMAT, CHANNELS, sizeInFrames, NULL, NULL, &aux1Buffer);
+        if (result != MA_SUCCESS) {
+            printf("Failed to initialize ring buffer.\n");
+            return result;
+        }
+        ma_silence_pcm_frames(aux1Buffer.rb.pBuffer, sizeInFrames, FORMAT, CHANNELS);
+        ma_writer_node_config writerConfig = ma_writer_node_config_init(CHANNELS, SAMPLE_RATE * 5, &aux1Buffer);
+        result = ma_writer_node_init(ng, &writerConfig, NULL, &m_sendToAux[id]);
+        if (result != MA_SUCCESS) {
+            cout << "ERROR " << result << ": Failed to init writer node." << endl;
+            return result;
+        }
+        result = ma_node_attach_output_bus(&m_sendToAux[id], 0, ma_engine_get_endpoint(&m_engine[id]), 0); // Pull API
+        if (result != MA_SUCCESS) {
+            cout << "ERROR " << result << ": Failed to attach writer node." << endl;
+            return result;
+        }
+    }
     while (result == MA_SUCCESS && i < m_layersQty) {
         result = this->createFilterBank(id, i);
         i++;
@@ -176,12 +238,19 @@ bool MiniAudioEngine::startDevice(uint *systemId, uint nb)
 
     m_devicesSelected = nb;
     for (uint internalId = 0; internalId < nb; internalId++) {
-        deviceConfig = ma_device_config_init(ma_device_type_playback);
+        deviceConfig = ma_device_config_init(ma_device_type_duplex);
+        deviceConfig.capture.pDeviceID  = &m_pPlaybackDeviceInfos[systemId[internalId]].id;
+        deviceConfig.capture.format     = m_resourceManager.config.decodedFormat;
+        deviceConfig.capture.channels   = CHANNELS;
+        deviceConfig.capture.shareMode  = ma_share_mode_shared;
         deviceConfig.playback.pDeviceID = &m_pPlaybackDeviceInfos[systemId[internalId]].id;
         deviceConfig.playback.format    = m_resourceManager.config.decodedFormat;
-        deviceConfig.playback.channels  = 0;
+        deviceConfig.playback.channels  = CHANNELS;
         deviceConfig.sampleRate         = m_resourceManager.config.decodedSampleRate;
-        deviceConfig.dataCallback       = audioDataCallback;
+        if (internalId == 0)
+            deviceConfig.dataCallback   = audioDataCallback;
+        else if (internalId == 1)
+            deviceConfig.dataCallback   = audioDataCallback1;
         deviceConfig.pUserData          = &m_engine[internalId];
         result = ma_device_init(&m_context, &deviceConfig, &m_device[internalId]);
         if (result != MA_SUCCESS) {
@@ -191,6 +260,7 @@ bool MiniAudioEngine::startDevice(uint *systemId, uint nb)
         engineConfig                  = ma_engine_config_init();
         engineConfig.pDevice          = &m_device[internalId];
         engineConfig.pResourceManager = &m_resourceManager;
+        engineConfig.gainSmoothTimeInMilliseconds = SAMPLE_RATE / 100;
         engineConfig.noAutoStart      = MA_TRUE;
         result = ma_engine_init(&engineConfig, &m_engine[internalId]);
         if (result != MA_SUCCESS) {
@@ -209,6 +279,23 @@ bool MiniAudioEngine::startDevice(uint *systemId, uint nb)
         }
         cout << "Initialized Audio Device. internalId: " << internalId << " systemId: " << systemId[internalId] << " " << m_pPlaybackDeviceInfos[systemId[internalId]].name << endl;
     }
+    //result = ma_data_source_rb_init(&g_dataSourceRB, &aux1Buffer);
+    if (result != MA_SUCCESS) {
+        cout << "Error " << result << ": Failed to init data source ring buffer" << endl;
+        return false;
+    }
+    ma_data_source_node_config dataSupplyNodeConfig = ma_data_source_node_config_init(&g_dataSourceRB);
+    //result = ma_data_source_node_init(ma_engine_get_node_graph(&m_engine[1]), &dataSupplyNodeConfig, NULL, &g_dataSupplyNode);
+    if (result != MA_SUCCESS) {
+        cout << "Error " << result << ": Failed to init data source node" << endl;
+        return false;
+    }
+    //result = ma_node_attach_output_bus(&g_dataSupplyNode, 0, ma_engine_get_endpoint(&m_engine[1]), 0);
+    if (result != MA_SUCCESS) {
+        cout << "Error " << result << ": Failed to attach data source rb node" << endl;
+        return false;
+    }
+    cout << "data source node state " << ma_node_get_state(&g_dataSupplyNode.base) << endl;
     return true;
 }
 
@@ -255,12 +342,8 @@ ma_result MiniAudioEngine::loadMedia(int layer, char *file, uint audioDevice)
 {
     ma_result result;
 
-    // ToDo: ver si s puede attach dos dispositivos a la vez. si no:
-    // - enchufar a un splitter al sonido y attach cada uno de los lados.
     // - iniciar un sonido por cada capa, copiar la capa en otro dispositivo
-    // - splitter al final de filterBank, esas señales se mezclan en un nodo mudo
-    // y se escribe la mezcla en un buffer. Mezclar el buffer en disco con el
-    // del otro device en el audio callback .
+    // - writer node y source con el buffer escrito en el otro device
     if (m_mediaLoaded[layer] == true)
     {
         ma_sound_uninit(&m_currentSound[layer]);
@@ -275,7 +358,7 @@ ma_result MiniAudioEngine::loadMedia(int layer, char *file, uint audioDevice)
     }
     result = ma_node_attach_output_bus(&m_currentSound[layer], 0, &m_filterBank[audioDevice][layer].input, 0);
     if (result != MA_SUCCESS) {
-        cout << "Error " << result << ": Failed to attach output bus " << audioDevice << endl;
+        cout << "Error " << result << ": Failed to attach sound output bus " << audioDevice << endl;
         //return result;
     }
     m_mediaLoaded[layer] = true;
@@ -343,7 +426,7 @@ void MiniAudioEngine::volChanged(int layer, float vol)
         return;
     if (vol >= 1)
         vol = 0.99f;
-    ma_sound_group_set_fade_in_milliseconds(&m_currentSound[layer], -1, vol, FADE_TIME);
+    ma_sound_group_set_fade_in_milliseconds(&m_currentSound[layer], -1, pow(vol, 3), FADE_TIME);
     m_currentLayerValues[layer].vol = vol;
 }
 
@@ -540,8 +623,20 @@ ma_result MiniAudioEngine::filterParamChanged(int layer, int audioDevice, int ch
             cout << "ERROR " << result << ": Failed set gain high shelf filter node." << endl;
             return result;
         }
-    }
-    return (result);
+    } else if (channel == SEND1) {
+        ma_node_set_output_bus_volume(&fb->output, 0, pow((value / 255.0f), 2));
+        if (result != MA_SUCCESS) {
+            cout << "ERROR " << result << ": Failed set Send 1 Volume." << endl;
+            return result;
+        }
+    }  else if (channel == SEND2) {
+        ma_node_set_output_bus_volume(&fb->output, 1, pow((value / 255.0f), 2));
+       if (result != MA_SUCCESS) {
+           cout << "ERROR " << result << ": Failed set Send 2 Volume." << endl;
+       }
+       return result;
+   }
+   return (result);
 }
 
 bool MiniAudioEngine::setBypass(int audioDevice, int layer, bool bypass)

src/miniaudioengine.h

@@ -1,21 +1,20 @@
 #ifndef MINIAUDIOENGINE_H
 #define MINIAUDIOENGINE_H
 
-#define MA_ENABLE_ONLY_SPECIFIC_BACKENDS 1
-#define MA_ENABLE_JACK 1
-#define MA_NO_GENERATION 1
-#define MA_DEBUG_OUTPUT 1
+#define MA_ENABLE_ONLY_SPECIFIC_BACKENDS
+#define MA_ENABLE_JACK
+#define MA_NO_GENERATION
+#define MA_DEBUG_OUTPUT
 #define MA_DISABLE_PULSEAUDIO
+#define MA_DEBUG_OUTPUT
+#define MA_LOG_LEVEL_DEBUG DEBUG
 #define MINIAUDIO_IMPLEMENTATION
 #include "miniaudio.h"
-#include "defines.h" // MAX_LAYERS
+#include "ma_writer_node.h"
 #include <bits/stdc++.h>
 using namespace std;
+#include "defines.h"
 
-/* Data Format */
-#define FORMAT              ma_format_f32   /* Must always be f32. */
-#define CHANNELS            2
-#define SAMPLE_RATE         48000
 
 typedef struct
 {
@@ -31,12 +30,23 @@ typedef struct
     ma_hishelf_node hishelf;
     ma_hishelf_node_config hishelfConfig;
     ma_splitter_node output;
+    ma_writer_node send1;
+    ma_data_source_node return1;
+    ma_audio_buffer_ref supplyReturn1;
 } filterBank;
 
+typedef struct
+{
+   ma_engine m_engine[MAX_AUDIODEVICES];
+   filterBank m_filterBank[MAX_AUDIODEVICES][MAX_LAYERS];
+   ma_writer_node m_sendAux1[MAX_LAYERS];
+   ma_pcm_rb *aux1Buffer;
+} audioObjects;
 
 class MiniAudioEngine
 {
     friend class libreMediaServerAudio;
+    static ma_pcm_rb *rb;
 
 public:
     MiniAudioEngine();
@@ -44,6 +54,7 @@ public:
     bool startEngine(uint layersQty);
     bool startDevice(uint *id, uint nb);
     static void audioDataCallback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
+    static void audioDataCallback1(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
     bool setBypass(int audioDevice, int layer, bool bypass);
 
 protected:
@@ -76,6 +87,8 @@ private:
     filterBank m_filterBank[MAX_AUDIODEVICES][MAX_LAYERS];
     ma_engine m_engine[MAX_AUDIODEVICES];
     uint m_layersQty;
+    ma_writer_node m_sendToAux[MAX_LAYERS];
+    audioObjects m_audioObjects;
 
     ma_result getAllAudioDevices();
     ma_result startContext();